Enterprises of all sizes are expanding Internet Protocol (“IP”) telephony and IP video deployments such that what is traditionally thought of as separate voice, video and data networks are converging to run over an IP infrastructure. Even though voice and video are sometimes characterized as just other applications, the fundamental aspects of voice/video conversations place requirements on the network that are quite different from data applications. These requirements amount to providing toll quality voice (and video), which is measured in terms of clarity and delay. Unlike data applications in which automatic retransmissions of erred data is expected and easily handled, there are no second chances with voice (and video).
In general, IP does not provide a mechanism to ensure that data packets are delivered in sequential order, or provide Quality of Service (“QoS”) guarantees, so Voice over IP (“VoIP”) and video over IP (also referred to as “IP streaming video”) implementations face problems dealing with latency, packet loss, and jitter. One type of latency problem resulting in network degradation is “absolute” or “fixed” delay that can cause a perceived loss of voice/video quality. A wide range of factors contribute to fixed delay including encoding delay from the chosen codec algorithm, switching time for each individual packet (also known as packet time), propagation time in the network and delay from optional encryption, intrusion detection filtering and similar processes.
Packet loss can be viewed as an extreme case of delay where the packets are so severely delayed that they never arrive. For example, if a network failure occurs, packets may be lost during the time that traffic is rerouted onto alternate facilities or for some OSI layer 2 protocols such as frame relay, Asynchronous Transfer Mode (“ATM”) and Multiprotocol Label Switching (“MPLS”), errant packets are detected and discarded.
Packet jitter is used to describe the difference between the longest delay and the shortest delay in the delivery of packets traversing the network, link or pathway during a predetermined period of time. Sometimes, packet jitter is used to describe the maximum delay difference between two consecutive packets in some period of time. For most data applications, this has a minor impact, as data protocols are designed to collect information and to transmit and receive this information whenever it is available. As long as each packet arrives intact, the timing between packets is of relatively minor importance. This is referred to as asynchronous transmission—there is no fixed relationship between the timing at the sending and the receiving end. Voice/video is quite different, as it is a synchronous service—which requires a more precise delay relationship between the source and the recipient of the information.
Additional supported voice and video problems include echo, one way voice path, gaps in speech and distorted/choppy voice/video. The above described problems may be caused by duplex mismatch, blocked IP packets (due to firewall or network address translation (“NAT”)), congestion, low-speed link in path, fast pipe to slow pipe, route flapping and/or link failure.
Existing voice and video quality management solutions are manual and therefore time-consuming, laborious and prone to error. These solutions require two or more different applications for alerts and performance data to locate a problem cause and at least one more application to apply a needed corrective policy. In addition, it is nearly impossible for a network administrator to continuously monitor network traffic and also take corrective action in a high availability network environment. Therefore, a need exists for call quality management systems and methods that can monitor a network system, determine the likely cause of the problem, locate the actual source of the problem and perform most corrective actions in an automated fashion to solve the aforementioned problems.